Chan_sip.c: Retransmission Timeout Reached On Transmission


Hello friends:

I am facing cutoffs randomly when negotiating calls.

The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]*

I hope anyone can give me an idea to solve this issue. Softswitch is using an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with Asterisk

Thanks in advance

Elder D. Arohuanca Lima - Peru

*[1]* [Aug 12…

Asterisk Users 15 days ago 3 Answers

Asterisk 13 FAX


Hello team! I’m planning to add fax functionality to my PBX. From research it seems that there is 2 options: spandsp and Digium. I lean towards Digium app, licensing is fine. However, they don’t have download for v13 Should I just download their version for v12 Asterisk? Any other suggestions on what to use, what works best? I have a pretty good plan on what I’m going to do but unsure which one to pick. Callcentric is my SIP provider and it supports T.38 Thanks Ivan --

Asterisk Users 2 months ago 2 Answers

Small Homebrew Pbx


Hello all,

I'm new here and I'm interested in building a small PBX with asterisk at home. I have one single PSTN line and ethernet cabling in place. I already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM and RAID 10 SATA disks). I make and receive 10 calls a day on average. I want 4 IP phones connected to the ethernet network. When there is a incoming call, all phones must ring and the first that takes the call makes the others stop ringing, but lets them available for internal calls.

Given the requirements above,…

Asterisk Users 3 months ago 9 Answers

Asterisk Proxying A REFER


Hello, we are using Asterisk with Adhearsion as our application server, with another Asterisk box acting as the office PBX, where all office phones are registered.

A REFER to transfer calls within the office results in the Adhearsion application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure Asterisk 11 to proxy a refer across a bridge instead of following it, so the application server can follow it instead?

Best regards,

Luca Pradovera

Asterisk Users 4 months ago 2 Answers

Asterisk 13. Writing Call Quality Parameters To CDR. How?



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Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11.

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Asterisk Users 5 months ago 2 Answers

Asterisk Executable Suddenly About 40KB


such as Nagios is create a service that checks the size of the binary every can also take the perf data and graph it using one of the many Nagios graphing tools available. You can even use something like Munin for a task like this. I couldn't get along without >this. On some PBX's I have, I monitor over 600 different metrics spread out every 1,5,10,15,30, and 60 minutes. Because they're spread out, the load average from these checks is zero. Just a suggestion.

Hi John

Thanks for the suggestions I'll keep them in mind.

Just to confirm back to the list, the…

Asterisk Users 8 months ago 0 Answers

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First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65.

I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC:

<[TK]D-Fender> Note for elfranne's situation, : nat=force_rport,comedia" should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it?

A call sample 202 calling 203 (ignore 403):

Asterisk Users 10 months ago 3 Answers

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OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this:Tier 1 Provider (SIP) > Asterisk 1.8 > Name Brand PBX - Calls work fine Outbound calls flow like this:Name Brand PBX > Asterisk 1.8 > Tier 1 provider (SIP) - Calls work fine

Problem is being reported on that many (not all) calls have no audio when they are forwarded. Example of forwarded call:Inbound call comes in from Tier 1 Provider…

Asterisk Users 11 months ago 2 Answers

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Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number.

For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ?

Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get those revenues, why a hacker would like to dial the same numbers over and over ? In other words, in this case, is looking at callee number a promising path to find hackers ?


Asterisk Users 11 months ago 15 Answers

SIP Fraud IP Blacklist



in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are:

To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply but attacks don't stop within a week

Definition of attack: - Minimum 5 attempts to make an unauthorized phone call to a non-PBX-internal number OR - Minimum 10 attempts to make an…

Asterisk Users 1.4 years ago 1 Answer